Inbound Routes

Good afternoon, I am trying to forward an incoming call from a provider, which sends it to server .201, to server .235. On server .201, I direct it to an extension (15112022), and I register the extension on .235 as a trunk. However, when I try to make a call, I receive the following notification:
trunk configuration on server .235

An interconnect between two PBX systems should almost always be a trunk on both ends.
If it’s an extension on .201, the destination will always be 15112022 so the .235 system can’t distinguish between different incoming numbers that may be routed to it. On outgoing, the .235 system would have no control over the caller ID to be sent.

Why did you choose to use an extension?

I don’t know what this means. Is this an error message displayed on a phone, or in a log somewhere? Is any other information included?

I am new to this, and I have a DID from the provider. Incoming calls come to the .201 server, and I direct them to the extension (15112022), which I register on the .235 server to route the call to the .235 server. If there is a simpler way, I would like to hear it.

Here is the configuration of my trunk on the .235 server:

But when the call reaches the .235 server, I get the following error:
[2024-07-08 04:02:02] WARNING[8113][C-00000016]: chan_sip.c:17266 check_auth: username mismatch, have <****>, digest has
[2024-07-08 04:02:02] NOTICE[8113][C-00000016]: chan_sip.c:26364 handle_request_invite: Failed to authenticate device “+3672244320” sip:+3672244320@*****.201;tag=as6778962e

Sorry, but what you have created is a mess. The call from the extension will have the original caller’s number (caller ID) in the From field, so of course it won’t match the username.

Unless you have some strong reason (please explain), delete this stuff and start over with a simple pjsip trunk on each end. If the systems are on the same LAN subnet and reasonably secure from the internet, you can set them up Registration None and Authentication None, with SIP Server set to the IP address of the other system. If these systems are using port 5060 for chan_sip, then set SIP Server Port to whatever the other system has for pjsip Port to Listen On.